A. Efforts to Interwork Internet Telephony and Wireless Telephony Systems
Two of the fastest growing areas of telecommunications are wireless mobile telephony and Internet telephony. Second and third-generation digital systems such as the Global System for Mobile communications (GSM), the Universal Mobile Telecommunications System (UMTS), and wideband CDMA are bringing new levels of performance and capabilities to mobile communications. Meanwhile, both the Internet Engineering Task Force's Session Initiation Protocol (SIP) and the International Telecommunications Union's H.323 enable voice and multimedia telephone calls to be transported over an Internet Protocol (IP) network. Subscribers to each of these networks need to be able to contact subscribers on the other. There is, therefore, a need to interconnect the two networks, allowing calls to be placed between them.
Some research has been performed investigating various aspects of interworking mobile communication systems with IP-based systems. The iGSM system allows an H.323 terminal to appear to the GSM network as a standard GSM terminal, so that a GSM subscriber can have his or her calls temporarily delivered to an H.323 terminal rather than a mobile device. Systems have been described for interworking GSM's in-call handover procedures with H.323. However, neither of these approaches solves the general interworking question: what is the best way for calls to be delivered and routed between the two networks?
As both mobile and Internet telephony are already designed to interconnect with the Public Switched Telephone Network (PSTN), the easiest way to interconnect them would be simply to use the PSTN as an intermediate link. This is, however, inefficient and suboptimal, as compared to connecting the networks by interworking the protocols directly, for a number of reasons.
First of all, routing calls via the PSTN can result in inefficient establishment of voice circuits. This is a common problem in circuit-switched wireless systems called “triangular routing,” as illustrated in FIG. 1. Because a caller's local switch 10 does not have sufficient information to determine a mobile's correct current location, the signaling must travel to an intermediate switch 12 which can locate the subscriber correctly.
This intermediate switch 12 can be far away from the caller 14 and the destination even if the two are located in a geographically close area. Since voice circuits are established at the same time as the call signaling message is routed, the voice traffic could be transported over a long, inefficient route. Note that there is an architectural difference here between the American mobile system based on ANSI 41 and the European systems based on GSM/UMTS MAP. In the American system, calls are always routed through a home mobile switching center, which is in a fixed location for each subscriber, so the voice traffic for all of the subscriber's calls travels through that switch. By contrast, GSM improves on this routing by sending calls through a gateway mobile switching center, which can be located close to the originating caller. However, there are some cases, such as international calls, where an originating PSTN switch does not have enough information to conclude that a call is destined for the GSM/UMTS network, and thus routes it to the subscriber's home country. Because there is no way for circuit paths to be changed once they have been established, the call's voice traffic travels first to the user's home country and only then to his or her current location.
In Internet telephony, by contrast, the path of a call's media (its voice traffic, or other multimedia formats) is independent of the signaling path. Therefore, even if signaling takes a triangular route, the media travels directly between the devices which send and receive it. Since each device knows the other's Internet address, the packets making up this media stream are sent by the most efficient routes that the Internet routing protocols determine.
As we interwork Internet telephony with mobile telephony, we would like to maintain this advantage. We can accomplish this by supporting a direct IP connection between mobile base stations and IP terminals. With PSTN signaling, this is not possible, so IP telephony signaling must be used to establish this connection.
Another motivation for direct connection between mobile and Internet telephony is to eliminate unnecessary media transcoding. The Real-Time Transport Protocol (RTP), the media transport protocol common to both H.323 and SIP, can transport almost any publicly-defined media encoding. Most notably, the GSM 06.10 encoding is implemented by many clients. If a GSM mobile device talks to an RTP-capable Internet telephone with an intermediate PSTN leg, the media channel would have to be converted from GSM 06.10 over the air, to uncompressed (μ-law or a-law) audio over a PSTN trunk, and then again (likely) to some compressed format over the RTP media channel. The degradation of sound quality from multiple codecs in tandem is well known, and multiple conversions induce unnecessary computation. A direct media channel between a base station and an IP endpoint allows, by contrast, communication directly using the GSM 06.10 encoding without any intermediate transcodings.
Finally, on a broader scale, an integrated architecture supporting Internet and mobile telephony will evolve naturally with the expected telecommunications architectures of the future. Third-generation wireless protocols will support wireless Internet access from mobile devices. New architectures such as Router for Integrated Mobile Access (RIMA) for Mobile Switching Centers (MSCs) are using IP-based networks for communications between MSCs and base stations. In the fixed network, meanwhile, IP telephony is increasingly becoming the long-haul transport of choice even for calls that originate in the PSTN. The direct connection between Internet telephony and mobile networks takes advantage of all these changes in architecture and allows us to build on them for the future.
B. Signaling and Transport Between Internet Telephony and Wireless Telephony Systems
The volume of traffic carried over packet networks, especially IP networks, has grown exponentially over the last decade. While this traffic has initially been generated by data applications, such as email and Web browsing, packet transport for voice and multimedia traffic is attractive to service providers because it will allow a single integrated network to be operated. There are many research and standards efforts underway to advance the use of IP technology for both voice transport and signaling for network control.
New packet networks supporting voice telephony must interwork with existing circuit switched networks for transport and signaling. For transport, media gateways are used to transform circuit interfaces to packet interfaces and vice versa. The interfaces to control these media gateways are being standardized in industry. For signaling, gateway controllers interwork call control protocols for Internet Telephony, such as H.323 and the SIP, with the PSTN protocols, such as the ISDN User Part (ISUP).
These changes have so far applied largely to the wired infrastructure, but are now spreading to the wireless networks as well. Currently deployed wireless networks, termed Second Generation (2G) networks, are using new packet air interfaces to support packet data applications. The next generation wireless networks, termed Third Generation (3G) networks, which are currently being standardized and trialed, will have higher speed packet air interfaces that support hundreds of Kbits/sec of traffic.
Service providers are interested in first using packet transport for voice inside the wired portion of their access networks to reduce operational costs, take advantage of statistical multiplexing, and move towards a single back-bone network capable of supporting voice and data applications. As third generation networks are deployed, the packet voice interfaces may be extended all the way to the mobile terminal.
There currently exist several widely used air interface standards for 2G systems, including those based on Time Division Multiple Access (TDMA), such as GSM and IS-136, and Code Division Multiple Access (CDMA), such as IS-95. For 3G systems standards are converging around UMTS and DMA2000. Each of these air interfaces has a corresponding interface defined between the radio access network and the network access switching equipment.
Within the wired access network, all systems currently use circuit switched technology for transporting user/mobile information, and variants of ISUP for call control. Two standard protocols are widely deployed for mobility management: Mobile Application Part (MAP) for GSM systems and IS-41 for non-GSM systems.
FIG. 2 shows a simplified GSM/UMTS network 20. The mobile terminals or mobile stations (MS) 22 access the network 20 through a radio called the Base Terminal Station (BTS) 24. The BTS terminates the air interface with the mobile terminals. Multiple BTSs connect to a Base Station Controller (BSC) 26. The BSC 26 manages handoffs between BTSs 24 and provides a common interface to the MSC 28 called the A-interface. Current MSCs are circuit switches that are responsible for mobility management, call control, service access, and user traffic switching. Mobility management includes registering and authenticating mobile devices, directing handoffs between BSCs, and paging to locate mobile terminals.
The MSC 28 includes an internal database called the VLR 21 which is used to store profiles for the mobile terminals it is currently serving. The MSC 28 interacts with other switches to manage calls through an ISUP interface, and network databases using MAP. The network databases, which store permanent copies of user profiles and keep track of their current location, are called Home Location Registers (HLR) 23. Both ISUP and MAP are part of the Signaling System no. 7 (SS7), the family of signaling protocols used in the PSTN 25.
From the discussion above, it becomes clear that to add packet transport to a cellular network, wireless access switching equipment must support multiple interfaces for signaling and transport. In essence, the media and signaling gateways now being developed for wired packet-based telephony systems must also be built for wireless networks, and MSCs must control these gateways and support the various radio standards. One major challenge to a wireless telephony system is that they must be able to handle traffic mixes that vary at a much higher degree than those for their wired counterparts. The traffic mix may include call and mobility related Requests, Short Messaging Service (SMS), and supplementary services. The performance of a system will vary depending on this traffic mix overall and the ratio of calls and mobility, called call-to-mobility ratio, in particular. The call-to-mobility ratio for urban settings, for example, may differ significantly from that in rural settings. Other aspects that may affect the traffic pattern include differences in countries/regions of deployment and varying cost structures, among others.
Service providers seek wireless mobile systems that support IP telephony, handle various air interfaces, and can be deployed in various settings. While many of these issues are well understood and have been addressed individually, building a system that addresses them in an integrated fashion is extremely challenging.